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* Store default speaker configurations in a structChris Robinson2014-10-021-11/+14
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* Make ComputeAngleGains use ComputeDirectionalGainsChris Robinson2014-10-022-5/+6
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* Don't use ComputeAngleGains for SetGainsChris Robinson2014-10-021-1/+5
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* Use an ambisonics-based panning methodChris Robinson2014-09-302-0/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For mono sources, third-order ambisonics is utilized to generate panning gains. The general idea is that a panned mono sound can be encoded into b-format ambisonics as: w[i] = sample[i] * 0.7071; x[i] = sample[i] * dir[0]; y[i] = sample[i] * dir[1]; ... and subsequently rendered using: output[chan][i] = w[i] * w_coeffs[chan] + x[i] * x_coeffs[chan] + y[i] * y_coeffs[chan] + ...; By reordering the math, channel gains can be generated by doing: gain[chan] = 0.7071 * w_coeffs[chan] + dir[0] * x_coeffs[chan] + dir[1] * y_coeffs[chan] + ...; which then get applied as normal: output[chan][i] = sample[i] * gain[chan]; One of the reasons to use ambisonics for panning is that it provides arguably better reproduction for sounds emanating from between two speakers. As well, this makes it easier to pan in all 3 dimensions, with for instance a "3D7.1" or 8-channel cube speaker configuration by simply providing the necessary coefficients (this will need some work since some methods still use angle-based panpot, particularly multi-channel sources). Unfortunately, the math to reliably generate the coefficients for a given speaker configuration is too costly to do at run-time. They have to be pre- generated based on a pre-specified speaker arangement, which means the config options for tweaking speaker angles are no longer supportable. Eventually I hope to provide config options for custom coefficients, which can either be generated and written in manually, or via alsoft-config from user-specified speaker positions. The current default set of coefficients were generated using the MATLAB scripts (compatible with GNU Octave) from the excellent Ambisonic Decoder Toolbox, at https://bitbucket.org/ambidecodertoolbox/adt/
* Combine some fields into a structChris Robinson2014-09-101-3/+6
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* Invert the ChannelOffsets arrayChris Robinson2014-09-101-1/+2
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* Remove the GetLatency method from the old BackendFuncsChris Robinson2014-09-081-4/+0
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* Convert the winmm backend to the new backend APIChris Robinson2014-09-081-3/+0
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* Make the fontsound's buffer and link fields atomicChris Robinson2014-09-033-17/+25
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* Protect alProcessUpdatesSOFT with a lockChris Robinson2014-09-031-2/+2
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* Use proper atomics for the thunk arrayChris Robinson2014-09-031-12/+13
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* Make the buffer's pack and unpack properties atomicChris Robinson2014-09-032-11/+11
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* Remove a couple unnecessary typedefsChris Robinson2014-08-241-3/+0
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* Convert the wave writer backend to the new APIChris Robinson2014-08-241-3/+0
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* Use al_malloc/al_free for default allocatorsChris Robinson2014-08-241-2/+2
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* Rename activesource to voiceChris Robinson2014-08-215-52/+51
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* Use an array of objects for active sources instead of pointersChris Robinson2014-08-213-48/+37
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* Use a NULL source for inactive activesourcesChris Robinson2014-08-214-36/+45
| | | | Also only access the activesource's source field once per update.
* Update COPYING to the latest ↵François Cami2014-08-1810-20/+20
| | | | https://www.gnu.org/licenses/old-licenses/lgpl-2.0.txt to fix the FSF' address Fix the FSF' address in the source
* ALC_SOFT_pause_device is finishedChris Robinson2014-08-121-10/+0
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* Disable the autowah effectChris Robinson2014-08-061-0/+2
| | | | | | | | There's apparently some issues with it causing noise or killing the output. It might be due to the per-sample changes being too harsh for the filter to keep up with, but it's not something I can take care of in time for release. This commit should be reverted after release when work on fixing it can resume.
* Fix some lock ordering to avoid potential deadlocksChris Robinson2014-08-031-8/+8
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* Use an ATOMIC_INIT macro instead of ATOMIC_LOAD_UNSAFEChris Robinson2014-08-032-4/+4
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* Add some casts for inline assembly atomicsChris Robinson2014-08-011-1/+1
| | | | And remove an unnecessary void cast
* Use atomics for the device and context list headsChris Robinson2014-08-011-2/+2
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* Make the source's buffer queue head and current queue item atomicChris Robinson2014-07-312-60/+72
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* Rename ATOMIC_COMPARE_EXCHANGE to ATOMIC_COMPARE_EXCHANGE_STRONGChris Robinson2014-07-311-1/+1
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* Always set the active source's update methodChris Robinson2014-07-261-5/+7
| | | | | | | If the source is stopped, changes its buffer, then played again quickly, the source will never be removed from the active source list causing the update method to remain as it was. If the buffer was changed between mono and multi- channel, this causes it to use the wrong method.
* Explicitly pass the address of atomics and parameters that can be modifiedChris Robinson2014-07-265-49/+49
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* Support C11 atomicsChris Robinson2014-07-231-1/+2
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* Use generic atomics in more placesChris Robinson2014-07-225-10/+10
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* Add macros for generic atomic functionalityChris Robinson2014-07-226-44/+44
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* Make some functions staticChris Robinson2014-07-202-6/+4
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* Load the default soundfont as a comma-separate list of filenameChris Robinson2014-07-191-6/+27
| | | | | | This allows multiple soundfont files to be "patched" together to create a single soundfont. For instance a GM soundfont with a separate soundfont for GS-only additions.
* Add a source radius property that determines the directionality of a soundChris Robinson2014-07-112-0/+4
| | | | | | | | | At 0 distance from the listener, the sound is omni-directional. As the source and listener become 'radius' units apart, the sound becomes more directional. With HRTF, an omni-directional sound is handled using 0-delay, pass-through filter coefficients, which is blended with the real delay and coefficients as needed to become more directional.
* Store 4 modulators per map entryChris Robinson2014-07-062-8/+14
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* Regroup and reorganize some macrosChris Robinson2014-07-061-40/+57
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* Use VECTOR_FIND_IF instead of a manual loopChris Robinson2014-07-061-11/+7
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* Allow ALsoundfont_deleteSoundfont to handle multiple buffersChris Robinson2014-07-061-10/+20
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* Don't require pre-declaring vector typesChris Robinson2014-07-062-11/+6
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* Avoid aliasing an int arrayChris Robinson2014-07-051-4/+4
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* Make some more functions staticChris Robinson2014-07-052-7/+6
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* Use a helper function to check valid MIDI controller inputsChris Robinson2014-07-042-4/+15
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* Make a function staticChris Robinson2014-07-041-2/+2
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* AL_SOFT_MSADPCM is functionally completeChris Robinson2014-07-031-6/+0
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* Simplify setting a fontsound linkChris Robinson2014-07-011-10/+6
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* Check that a fontsound is NOT null before deleting itChris Robinson2014-07-011-1/+1
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* Remove an unused variableChris Robinson2014-07-011-2/+0
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* Standardize some New/Delete methodsChris Robinson2014-06-306-27/+32
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* Remove an unused macroChris Robinson2014-06-291-1/+0
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